Music, Programming, and other topics related to the modern Music Engineering Scene.

Microphones are analog by nature. The membrane physically interacts with the airwaves around it, recording any vibrations is senses. So how in the world do you make a digital microphone? These were my thoughts exactly as I entered the workshop on the development of these mics.

What the heck is a digital microphone?Before going too far into how digital microphones work, let's look at the regular analog flow of things.Signal goes from the microphone to a preamp through an XLR cable. From there, it goes through an analog to digital converter, then routed to the mixing board or whatever interface you're using for your computer. This is all well and good, except it could be better.

Microphones have a Signal to Noise Ratio (SNR) of about 138dB. 24 bit recordings can handle SNR of over 140dB. The preamp, however, has a SNR of around 100dB. You lose 40dB of dynamic range because of this external preamp setup. How do you get around this problem? Digital microphones propose the solution.

Digital microphones combine these three components into one. The microphone has an on-board preamp and A/D converter, giving you a digital signal out instead of the analog signal. It uses special quantizing processes to get the full dynamic range of the microphone and eliminate the 100dB limit of usual preamps. The actual methods are a little too technical for this post - I may come back and dedicate a post to that process.

Great, so now I have to buy an entirely new mic collection?Not so fast, there. The spec for digital microphones (AES42, I think)is still being developed. Even after it's finalized, there's no reason to replace mics that sound killer. Digital mics are meant to augment what's currently available, not completely replace it.

What problems can I expect?Ideally, none. Digital mics should be ready to go. Unfortunately, any experienced audio engineer can tell you that with digital signals comes a headache of problems. What about sample rates? Who controls the mic's sample rate? What sort of interfaces will be needed for these signals?These are all problems that are being worked out.

There are currently 2 modes of operation for digital mics. Mode 1 specifies that the microphone is its own master. It only operates at the sample rate that it is made for, and that's that. Mode 2 requires a separate interface to dictate what sample rate the mic should operate at. Even with these distinctions, however, different companies still have slightly different ways of doing things - meaning lots of hardware won't interact well with other hardware. This will hopefully become more standardized as AES42 is developed.

That's the brief overview. As I mentioned earlier, I may come back to fill in details on some of these things in later posts. If you can't wait that long (or for more detailed information), look up the spec at www.hauptmikrofon.de/aes42/.

My apologies for the nine month hiatus. School has been keeping me busy. I'll try to post some cool interesting things over the next month, starting with crazy stuff from AES.

Too much went on during AES to list it all in one post. For those who have never been to the AES conference, it's basically a gathering of all the people who matter in the audio industry. The biggest names in microphones, software developers, hardware, and anyone else you can think of gather for four days of sheer madness. Lots of workshops, technical presentations, and an auditorium full of companies showing off their latest and greatest products. It's definitely something worth going to at least once in your life.

If you've ever used a DAW (Digital Audio Workstation, such as Pro Tools or Logic), you've probably heard of dither. You might have even used it without really knowing what it is or why it works. Put simply, Dither is very low level noise that is added to your signal to help quantize it more accurately.

Consider this signal:Note that the blue dots represent sample values. Since the signal does not go past halfway between bit values, we completely lose any trace of the signal. If we add some noise to this signal, we actually make it's representation slightly more accurate:By adding just a little bit of high frequency noise, we have successfully pushed the signal level past the half-way point, causing the system to round up to the next highest bit value. This preserves our original signal, with only a very small audible increase in volume.

You may have noticed that there are many different types of dither. You will commonly use dither to convert between different bit depths, and there may be a few different settings to choose from. The only difference between the settings is the noise content. All the settings are still noise, but many companies are spending quite a bit of time and money on shaping the noise so that it works the best without the ear being able to hear it. You can imagine how poorly designed dither algorithms could completely ruin a track, so having a good dither is very important.

As mentioned at the end of the last post, aliasing can be a problem when recording digitally. Aliasing occurs when you try to record frequencies that are too fast. You have a limit on which frequencies you can record based on your sample rate. The Nyquist Theorem says that you can only accurately record frequencies that are less than half of your sample rate. For a 44,100 Hz sample rate, you can record frequencies up to 22,050 Hz. This covers everything that human ears can hear.

When you try to record something greater than 22,050 Hz, you will get aliasing. This creates high pitched "chirpies" that don't actually exist, but get reproduced because of how samples are taken. Consider the following picture:The black sine wave is the original signal. The blue dots are the samples that get taken, and the green dashed line is the false frequency that gets reproduced. You can see it is a much lower frequency than the original signal, and is not at all what we wanted to capture.

To get around aliasing, we apply anti-aliasing filters. Basically, this is a low-pass filter with a cutoff frequency just below your Nyquist frequency (half of your sampling rate). This prevents any frequencies that are too high from ever trying to get recorded, protecting the signal that we can capture from unwanted aliasing.

With the advent of computers, audio has been digitized. This should not come as a shock to you, as we've been using digital audio for many years now. But just how is audio digitized? The next few posts go over digital audio - the good parts and the bad.

Digital Audio relies on Quantization and Sampling. Quantization is the act of taking the peaks of an audio signal and storing them as binary values. These values are accurate to whatever bit depth you're working in (16 bit for CDs). Sampling is the act of splitting your signal into little chunks, called samples. You record the values at a specified interval, called the sampling rate, so you can reproduce the signal later. The sample rate for CDs is 44,100 Hz, which means 44,100 samples are taken per second. This might seem like a huge task, but don't worry. Computers are really quick.

Think of this like a grid being superimposed over your signal:The horizontal grey lines are akin to your bit depth. The vertical lines are the sampling rate. The blue dots represent the samples that you would take of this wave form. You can see that they don't perfectly match up, but they're close enough that you can reproduce the wave form and still get something out that sounds similar to what you recorded.

Next up, we'll go over the limits of the digital model (aliasing), and ways around it.

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Chemzoar does not intend to provide complete walkthroughs of everything covered on this website. Before undertaking any projects proposed by this website, be sure to do your own research on top of what's listed here. Chemzoar does not take any responsibility for decisions made based on the content of this site.